I know that an ACIDized wave file is saved for use in Acid Pro by Sony.
You can stretch it without changing the pitch.

Okay I opened the screen for you now.
LOOPING MODE: Looping mode only applies to audio formats (.wav, .mp3 and .ogg) and determines how any decaying sound after the last bar of your project/loop is rendered. For example, the tail of a reverb of a sound may be important for the impression of smooth continuity when looping, or to prevent the decaying reverb in a 'straight' render being chopped off. If, after rendering the last bar from the song there is still sound decaying, this option sets how FL Studio should proceed. Leave remainder is the default. NOTE: If you are making loop files use .wav format, .mp3 in particular leaves a small silence at the start of the sound that will interfere with looping.

- Leave remainder - Expands the song length to capture any decaying sounds. If 'Leave remainder' still chops of any trailing audio, the Playlist Repeat marker can be used to define the rendering end-point. While Repeat marker positions are usually ignored, if they are placed after the last Pattern, Audio or Automation Clip in the Playlist, the project will be rendered up to the position of the Repeat marker.
- Wrap remainder - Wraps any decaying sound at the end of the song onto the beginning (useful when rendering loops with effects). NOTE: This feature works by starting the render at the last bar, then mixing any audio decaying after the last bar back into the start of the song. If the decaying sound comes from notes before the last bar it won't be wrapped.
- Cut remainder - Cuts the render at the end of the last bar.

QUALITY: Sample Interpolation - Select the waveform interpolation method used for Sampler/Audio-Clip channels. Interpolation is a curve fitting process that computes intermediate sample amplitude data between the known sample points (filling in the gaps). This is only required when samples are transposed from their original pitch and the program calls for a sample value out of sync with the source data-points. Without interpolation quantizing (amplitude) errors can create unwanted high-frequency harmonic artifacts (aliasing & quantizing errors).
- Linear interpolation is the fastest method. It provides basic linear averaging between samples, however it can result in aliasing (high frequency noises) if samples are transposed far from their original pitch.
- 6-point Hermite has been optimized to be a quick curve interpolation method with superior quality to linear interpolation. It is ideal for exporting 'working drafts' of your audio files.
- 64, 128, 256, 512-point Sinc methods provide increasing quality interpolation, but they are also very slow. We recommended that you use at least 64-point Sinc on your final render, or better still, the maximum Sinc value that you are prepared to wait to finish rendering.

- Dithering - Applies 32 to 16-bit dithering to 16-bit .wav and .mp3 files. Dither should only be applied once to your final 16 bit audio file during the final render. If you plan to master or post-process your track then don't apply dithering.What is dithering? Dither breaks up the predictability, and so signal-like quality, of quantizing noise (rounding errors in signal amplitude that occur when transposing bit-depths, 32 to 16 for example) making it sound more like background hiss and less like an audio signal that will draw your attention (i.e. less noticeable). Quantizing noise is generally only audible in very quiet parts of a recording, where the music is approaching the limits of the bit-depth. Meaning, only a few bits are being used to represent the amplitude of the waveform. If you are hearing noise artifacts and your track is approaching 0 dB, then it is some other type of noise, probably aliasing (see the interpolation settings) or compression artifacts (mp3/ogg bit-depth).

- Alias-free TS404 - When enabled, prevents TS404 from "aliasing", but also slows down the rendering process.
- HQ for all plugins - Sets high quality mode for any plugins (effects and instruments) used in the song.
- Disable Max Poly - Ignores the max poly setting in Miscellaneous Channel Settings but does NOT ignore Mono option if selected.

WAVE: Wave is a lossless audio format and preferred for handling audio in a production environment (use it to save all your samples, sounds and archive material). The drop-down menu contains bit-depth options for the exported wave file -

- 16-Bit int wave is the highest-quality audio file compatible with a wide range of playback devices. CD audio format: If you want to create audio files compatible with CD format use 44.1 kHz, 16-bit .wav files. Check that the Mixer sample rate is set to 44.1 kHz in the Audio Settings window. Also note that FL Studio does not burn to CD format, it creates audio files ready for burning. Use any 3rd party CD burning program to create the audio CD.
- 24-Bit int wave is a common bit-depth used by DAW hardware & some older software DAWs. Use this bit-depth only if 32-Bit float is not supported by the 3rd party software/hardware.
- 32-Bit float wave is the native format of FL Studio mix engine. Render to 32-Bit floating point format when you intend to continue mixing or editing the file in another application (wave editor or DAW) that supports 32-Bit floating point format. 32-Bit float provides more precision for mathematical operations on the audio and so will ensure the highest quality is preserved in your audio files during post production activities.

OPTIONS:
- Split Mixer Tracks - When selected, each Mixer track in the project is exported as a separate .wav file. NOTE: this option does not export to mp3/ogg formats.
- Save ACIDized - Saves project tempo and slice/region markers in .wav/.mp3 files in Sony ACID™ meta-data format. Useful for programs that can read this data type. NOTE You can change the tempo using the Edison Sample Properties dialog and the slice/region markers in Edisons wave-edit window.
- Save Slice Markers - If enabled, each note will create a slice marker in the exported file. This means that FL Studio exports sliced drum loops which are automatically ready for slice re-ordering and high quality time stretching.
- Delay compensation - Applies only to .wav files and relates to PDC induced delays. When selected this strips PDC delay from the rendered .wav. Use this when start/end-timing is critical in the rendered .wav (when making loops for example). When deselected, a period of silence equal to the Master channel PDC will be added to the start of the .wav and the same again removed from the end. See also: Latency compensation this removes delay caused by the ASIO Buffer length setting during recording. If you have already recorded the audio, it's too late, you will need to manually align it to the Playlist if it has a delay induced offset. Sometimes unwanted rendering delays can be caused by Plugins Behaving Badly.